95 lines
2.6 KiB
C++
95 lines
2.6 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "modules/audio_processing/agc/agc.h"
|
|
|
|
#include <cmath>
|
|
#include <cstdlib>
|
|
#include <vector>
|
|
|
|
#include "modules/audio_processing/agc/loudness_histogram.h"
|
|
#include "modules/audio_processing/agc/utility.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
const int kDefaultLevelDbfs = -18;
|
|
const int kNumAnalysisFrames = 100;
|
|
const double kActivityThreshold = 0.3;
|
|
|
|
} // namespace
|
|
|
|
Agc::Agc()
|
|
: target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
|
|
target_level_dbfs_(kDefaultLevelDbfs),
|
|
histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
|
|
inactive_histogram_(LoudnessHistogram::Create()) {}
|
|
|
|
Agc::~Agc() = default;
|
|
|
|
void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
|
|
vad_.ProcessChunk(audio, length, sample_rate_hz);
|
|
const std::vector<double>& rms = vad_.chunkwise_rms();
|
|
const std::vector<double>& probabilities =
|
|
vad_.chunkwise_voice_probabilities();
|
|
RTC_DCHECK_EQ(rms.size(), probabilities.size());
|
|
for (size_t i = 0; i < rms.size(); ++i) {
|
|
histogram_->Update(rms[i], probabilities[i]);
|
|
}
|
|
}
|
|
|
|
bool Agc::GetRmsErrorDb(int* error) {
|
|
if (!error) {
|
|
RTC_NOTREACHED();
|
|
return false;
|
|
}
|
|
|
|
if (histogram_->num_updates() < kNumAnalysisFrames) {
|
|
// We haven't yet received enough frames.
|
|
return false;
|
|
}
|
|
|
|
if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
|
|
// We are likely in an inactive segment.
|
|
return false;
|
|
}
|
|
|
|
double loudness = Linear2Loudness(histogram_->CurrentRms());
|
|
*error = std::floor(Loudness2Db(target_level_loudness_ - loudness) + 0.5);
|
|
histogram_->Reset();
|
|
return true;
|
|
}
|
|
|
|
void Agc::Reset() {
|
|
histogram_->Reset();
|
|
}
|
|
|
|
int Agc::set_target_level_dbfs(int level) {
|
|
// TODO(turajs): just some arbitrary sanity check. We can come up with better
|
|
// limits. The upper limit should be chosen such that the risk of clipping is
|
|
// low. The lower limit should not result in a too quiet signal.
|
|
if (level >= 0 || level <= -100)
|
|
return -1;
|
|
target_level_dbfs_ = level;
|
|
target_level_loudness_ = Dbfs2Loudness(level);
|
|
return 0;
|
|
}
|
|
|
|
int Agc::target_level_dbfs() const {
|
|
return target_level_dbfs_;
|
|
}
|
|
|
|
float Agc::voice_probability() const {
|
|
return vad_.last_voice_probability();
|
|
}
|
|
|
|
} // namespace webrtc
|